In communication applications where channel bandwidth is at a premium, it is essential to use the smallest possible portion of a transmission channel in order to transmit a voice signal. A common solution is to process the voice signal with an apparatus called a speech codec before it is transmitted on a RF channel.
Speech codecs, including an encoding and a decoding stage, are used to compress (and decompress) the digital signals at the source and reception point, respectively, in order to optimize the use of transmission channels. By encoding only the necessary characteristics of a speech signal, fewer bits need to be transmitted than what is required to reproduce the original waveform in a manner that will not significantly degrade the speech quality. With fewer bits required, lower bit rate transmission can be achieved
Most state-of-the-art codecs are based on the original CELP odel proposed by Schroeder and Atal in "Code-Excited Linear Prediction (CELP): High Quality Speech at Very Low Bit Rates," Proceedings of ICASSP, pp. 937-940, 1985. This document is hereby incorporated by reference. This basic codec model has been improved in many aspects to achieve bit rates of approximately 8 kbits/sec and even lower, but voice quality in those with lower bit rates may not be acceptable for telephony applications. An example of an 8 kbits/sec codec is fully described in version 5.0 of the International Telecommunication Union Telecommunications Standardization Sector (ITU-TSS) Draft recommendation G.729 "Coding of speech at 8 kbits/s using Conjugate-Structure Algebraic-Code-Excited Linear-Predictive (CS-ACELP) coding", dated Jun. 8, 1995. This document is hereby incorporated by reference.
Considering that lower bit rates at acceptable speech quality provide great economical advantages, there exists a need in the industry to provide, an improved speech coding apparatus and method particularly well suited for telecommunications applications